[Freeswitch-users] Codec Negotiation: Codec telephone-event:101:8000:20

[Brian West]

NO the codec map includes 2 aka g726-32 which you don't have to list in the map if its defined in the standard... what I need is a pcap of the whole process.

/b

On Jan 6, 2010, at 6:43 PM, Mark Campbell-Smith wrote:

> Thanks guys for your response.  I'll have a read through the links sent to me.
> 
> On the INVITE my FS box sends to Phonzo (the SIP provider) the codecs
> it supports are G726-32, PCMU, PCMA.
> 
> v=0
> o=FreeSWITCH 1262742568 1262742569 IN IP4 xxx.xxx.xxx.xx
> s=FreeSWITCH
> c=IN IP4 xxx.xxx.xxx.xxx
> t=0 0
> m=audio 31566 RTP/AVP 2 0 8 101 13
> a=rtpmap:2 G726-32/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
> 
> However, Phonzo responds with the following in the Session Progress message:
> 
>  v=0
>  o=Sippy 158698636 1 IN IP4 80.232.37.178
>  s=-
>  t=0 0
>  m=audio 61812 RTP/AVP 2 101 13
>  c=IN IP4 213.50.91.3
>  a=rtpmap:101 telephone-event/8000
>  a=fmtp:101 0-15
> 
> Does that make sense?   The codec rtpmap only included DTMF ...